similar to: Queue App - Set monitoring dynamically

Hi all In my application I''ve some named routes defined this way... map.label_context1 '':context1/label'', :controller => ''mycontroller'' map.label_context2 '':context1/:context2/label'', :controller => ''mycontroller'' map.label_context3 '':context1/:context2/:context3/label'', :controller =>

Hi, I have a little situation with my dialplan, and I am wondering if what I want is even possible. Here it is: I have three contexts, context1 includes contexts2, and context2 includes context3. In other words, in context1 all extensions of context2 and context3 are valid (and actually working, so that's good). I am using those context for the sake of code clarity and reuse, and for

Hi all, is there an easy way to get a 'copy' of a type living in another context into the local context? Background: when calling a function residing in a different module (context2) from a module (context1), we first need to introduce a function declaration of the function with empty body. However, in order to do so, we need the function type. pFuncInContext2->getType gives us the

when i wnat to working with realtime and mysql for any context i have to insert (switch => Realtiem/context at extensions) statment into extensions.conf for example if i want to have 10 context, i have to insert these lines into extension.conf : [context1] switch => Realtiem/context1 at extensions [context2] switch => Realtiem/context2 at extensions [context3] switch =>

Hello, Is it possible to use the switch => statement in extensions.conf (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions) to point to a database and in the database use the include-statement ? In extconfig.conf I would have : extensions => mysql,asterisk,extensions_table In extensions.conf I would then have : [includecontext] switch => Realtime/includecontext at

Hi, I had the following odd behaviour in Asterisk 1.2 - We are migrating to 1.6, and I will re-test ASAP, though it is quite hard to replicate, but I am curious to know whether it is a known IAX issue in 1.2. We had 2 users in iax.conf: [user1] username=user1 secret=secret1 context=context1 host=iax.hostname.com [user2] username=user2 secret= context=context2 host=dynamic deny=0.0.0.0/0.0.0.0

I have several of incoming numbers on IAX from voiptalk and magrathea but have a problem with IAX.conf. If I follow the example from voiptalk [VoIPTalk Incoming Number] type=friend username=VoIPTalk Incoming Number context=[XXXXXXXX] and make incoming entries in IAX.conf for the numbers like below with a different entry for each number pointing to a different context, incoming numbers always

I am running Asterisk 1.8.10.1. I am trying to set up some routing in my dial plans and having some issues (the issue being that I don't quite understand all of the syntax and patterns that can be used: Examples: DID1 = 6140000000 DID2 = 6140000001 CNAME1 = 6149999999 CNAME2 = 6149999998 CNAME3 = 6149999997 context1 context2 context3 I have looked at several examples (patterns) and I

Hello list, I have this strange problem whenever I try to make an "ael reload" from the Asterisk CLI. The command gives the following result and crashes: voip-1*CLI> ael reload Disconnected from Asterisk server Executing last minute cleanups Asterisk ending (0). root at voip-1:/etc/asterisk# As far as I can see, aelparse can't find any errors in my configuration, following

Hi List, Just one more question that may sounds stupid to some people but I can't find the solution for now, I have the following dialplan: exten => queue,n,Queue(myqueue) exten => queue,n,NoOp(ENDQUEUE) I don't understand why the NoOp is never triggered, the incoming call is always hangup when the agent hangup... Is this a behaviour we can't get rid off without patching

I have a question about contexts calling each other. We have one * box that is setup for multiple companies. Calls come into the default context and that hands them out to the context for each company. For example, 1x goes to context1, 2x goes to context2, etc. Each context includes "outbound" which says that if you dial 1+ or a local number, you are sent out to the Cisco

Hello, I am running Asterisk 11 on CentOS 6.x. I have configured several queues as follows in extensions.conf: exten => s,1,Queue(myqueue,rtnC,18) same => n,Background(user_unavail) same => n,WaitExten(10) exten => 1,1,Voicemail(1111 at my-vm,s) This rings the phones in the queue for 18 seconds. If no queue members answer, the caller is then prompted to press 1 and leave a

Hi all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one

Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue "myqueue" In extension file: Queue(myqueue|t|||120) And my agents are joining in following

hi list, i have a question about the args of queue: when we use Queue() app, there are some arguments than can use. help from CLI: Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule]]]]]]]]) well.. i'm trying to identify who has taken the call on a queue, and, when agent conected, launch a macro with some args based on who takes the call i do: exten =>

Does anyone know of a way to specify what extension is dialed when 0 is pressed in the voicemail system. I have a situation where there is more than one secretary and they want the 0 to redirect to the appropriate secretary for the two groups of people. So an example would be: 555-1234 -> voicemail -> Secretary 1 555-1235 -> voicemail -> Secretary 2 Any help would be greatly

<!DOCTYPE html> <html><head> <meta charset="UTF-8"> </head><body><p>Hello everybody,<br><br>I manage not to detect one silence with record () when I make as follows:<br><br>Exten = > 0178900271, n, Record ($ ${ link_recorded_pseudos_clients } pseudo_ Client_Id} wav, 5,5) exten = > 0178900271, n, GotoIf ($ ["

Hello list, when I sent an incoming call first to a queue and after the timeout to a dial-command, while the correspondent's phone rings there is no ringtone for the caller... So it goes like this : 1. dial(SIP/account1,20) 2. queue(myqueue,,,,20) 3. dial(SIP/account2) In step 1 there is a ringtone for the caller. In step 2 there is musiconhold (class default) for the caller. In step 3

I have some Digium licensed Digium codecs, but when making a call and transcoding the call is only heard in one direction? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

Hi, I'm testing Asterisk 1.4.15 with the -g option. When it crash didn?t generate core file in the /tmp folder. What is happening?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071130/dc693449/attachment.htm